[asterisk-users] pjsip direct_media=yes and "unknown" endpoints

[Daniel Tryba]

On Wed, Apr 26, 2017 at 06:25:43PM +0200, Daniel Tryba wrote:
> Whoever when a terminating call comes in from the uplink provider, a
> sip request is send to a redirector. The redirector has
> redirect_method=uri_core configured (the only method that works for
> me).

[...]

> The request now gets routed based on a fully qualified domainname
> (with NAPTR/SRV records), which ultimately resolves to an ip that is
> matched in the endpoint SBC used above to originate a call.  But now
> the asterisk stays in the loop regarding RTP, a simple bridge is
> created but never switches to direct media.

This is not an asterisk problem, after fiddling with the config and
using templates to make sure the config for all (configured) endpoints
was the same, a reINVITE renegotiated RTP between the endpoints. 

However what happens is that after the renegotiation the DTMF
payloadtype (rfc4733) changed from 101 in the initial setup to 96. The
uplink provider doesn't support this thus DTMF breaks making
direct_media not an option right now. Something I have to figure out
later.